[Unified Plan] Don't end audio tracks when SSRC changes.

The RemoteAudioSource has an AudioDataProxy that acts as a sink, passing
along data from AudioRecvStreams to the RemoteAudioSource. If an SSRC is
changed (or other reconfiguration happens) with SDP, the recv stream and
proxy get recreated.

In Plan B, because remote tracks maps 1:1 with SSRCs, it made sense to
end remote track/audio source in response to this. In Plan B, a new
receiver, with a new track and a new proxy would be created for the new

In Unified Plan however, remote tracks correspond to m= sections. The
remote track should only end on port:0 (or RTCP BYE or timeout, etc),
not because the recv stream of an m= section is recreated. The code
already supports changing SSRC and this is working correctly, but
because ~AudioDataProxy() would end the source this would cause the
MediaStreamTrack of the receiver to end (even though the media engine
is still processing the remote audio stream correctly under the hood).

This issue only happened on audio tracks, and because of timing of
PostTasks the track would kEnd in Chromium *after* promise.then().

This CL fixes that issue by not ending the source when the proxy is
destroyed. Destroying a recv stream is a temporary action in Unified
Plan, unless stopped. Tests are added ensuring tracks are kLive.

I have manually verified that this CL fixes the issue and that both
audio and video is flowing through the entire pipeline:

Bug: chromium:1121454
Change-Id: Ic21ac8ea263ccf021b96a14d3e4e3b24eb756c86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214136
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33645}
7 files changed
tree: f45d2b762571dc61f17cc0380000b87955fc9201
  1. .clang-format
  2. .git-blame-ignore-revs
  3. .gitignore
  4. .gn
  5. .vpython
  7. BUILD.gn
  9. DEPS
  13. OWNERS
  15. PRESUBMIT.py
  16. README.chromium
  17. README.md
  19. abseil-in-webrtc.md
  20. api/
  21. audio/
  22. build_overrides/
  23. call/
  24. codereview.settings
  25. common_audio/
  26. common_video/
  27. data/
  28. docs/
  29. examples/
  30. g3doc.lua
  31. g3doc/
  32. license_template.txt
  33. logging/
  34. media/
  35. modules/
  36. native-api.md
  37. net/
  38. p2p/
  39. pc/
  40. presubmit_test.py
  41. presubmit_test_mocks.py
  42. pylintrc
  43. resources/
  44. rtc_base/
  45. rtc_tools/
  46. sdk/
  47. stats/
  48. style-guide.md
  49. style-guide/
  50. system_wrappers/
  51. test/
  52. tools_webrtc/
  53. video/
  54. webrtc.gni
  55. webrtc_lib_link_test.cc
  56. whitespace.txt

WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.

Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.

The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.


See here for instructions on how to get started developing with the native code.

Authoritative list of directories that contain the native API header files.

More info